Delete class SSRCDatabase, and its global ssrc registry.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 93 matching lines @@
       rtp_stats_callback_(nullptr),
       total_bitrate_sent_(kBitrateStatisticsWindowMs,
                           RateStatistics::kBpsScale),
       nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
       send_packet_observer_(send_packet_observer),
       bitrate_callback_(bitrate_callback),
       // RTP variables
-      ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
       remote_ssrc_(0),
       sequence_number_forced_(false),
-      ssrc_forced_(false),
       last_rtp_timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
       media_has_been_sent_(false),
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
       rtp_overhead_bytes_per_packet_(0),
       retransmission_rate_limiter_(retransmission_rate_limiter),
       overhead_observer_(overhead_observer),
       send_side_bwe_with_overhead_(
           webrtc::field_trial::FindFullName(
               "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
-  ssrc_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_ != 0);
-  ssrc_rtx_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_rtx_ != 0);
-
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
 
   // Store FlexFEC packets in the packet history data structure, so they can
   // be found when paced.
   if (flexfec_sender) {
     flexfec_packet_history_.SetStorePacketsStatus(
         true, kMinFlexfecPacketsToStoreForPacing);
   }
 }
 
 RTPSender::~RTPSender() {
   // TODO(tommi): Use a thread checker to ensure the object is created and
   // deleted on the same thread.  At the moment this isn't possible due to
   // voe::ChannelOwner in voice engine.  To reproduce, run:
   // voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
 
   // TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
   // variables but we grab them in all other methods. (what's the design?)
   // Start documenting what thread we're on in what method so that it's easier
   // to understand performance attributes and possibly remove locks.
-  if (remote_ssrc_ != 0) {
-    ssrc_db_->ReturnSSRC(remote_ssrc_);
-  }
-  ssrc_db_->ReturnSSRC(ssrc_);
-
-  SSRCDatabase::ReturnSSRCDatabase();
   while (!payload_type_map_.empty()) {
     std::map<int8_t, RtpUtility::Payload*>::iterator it =
         payload_type_map_.begin();
     delete it->second;
     payload_type_map_.erase(it);
   }
 }
 
 uint16_t RTPSender::ActualSendBitrateKbit() const {
   rtc::CritScope cs(&statistics_crit_);
@@ skipping 151 matching lines @@
   rtx_ = mode;
 }
 
 int RTPSender::RtxStatus() const {
   rtc::CritScope lock(&send_critsect_);
   return rtx_;
 }
 
 void RTPSender::SetRtxSsrc(uint32_t ssrc) {
   rtc::CritScope lock(&send_critsect_);
-  ssrc_rtx_ = ssrc;
+  ssrc_rtx_.emplace(ssrc);
 }
 
 uint32_t RTPSender::RtxSsrc() const {
   rtc::CritScope lock(&send_critsect_);
-  return ssrc_rtx_;
+  RTC_DCHECK(ssrc_rtx_);
+  return *ssrc_rtx_;
 }
 
 void RTPSender::SetRtxPayloadType(int payload_type,
                                   int associated_payload_type) {
   rtc::CritScope lock(&send_critsect_);
   RTC_DCHECK_LE(payload_type, 127);
   RTC_DCHECK_LE(associated_payload_type, 127);
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
     return;
@@ skipping 41 matching lines @@
                                  size_t payload_size,
                                  const RTPFragmentationHeader* fragmentation,
                                  const RTPVideoHeader* rtp_header,
                                  uint32_t* transport_frame_id_out) {
   uint32_t ssrc;
   uint16_t sequence_number;
   uint32_t rtp_timestamp;
   {
     // Drop this packet if we're not sending media packets.
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_) {
+      LOG(LS_ERROR)  << "SSRC unset";

[stefan-webrtc, 2017/02/15 08:26:37]

End with .

Same for the logs below.

Should we also DCHECK? Seems like incorrect usage to me.

[nisse-webrtc, 2017/02/15 08:42:03]

Done for all log messages in this file.
Maybe. I'm aiming for making the ssrc a construction time const, but we're not
there yet.

+      return false;
+    }
+
+    ssrc = *ssrc_;
     sequence_number = sequence_number_;
     rtp_timestamp = timestamp_offset_ + capture_timestamp;
     if (transport_frame_id_out)
       *transport_frame_id_out = rtp_timestamp;
     if (!sending_media_)
       return true;
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
     LOG(LS_ERROR) << "Don't send data with unknown payload type: "
@@ skipping 99 matching lines @@
       if (rtx_ == kRtxOff) {
         if (payload_type_ == -1)
           break;
         // Without RTX we can't send padding in the middle of frames.
         // For audio marker bits doesn't mark the end of a frame and frames
         // are usually a single packet, so for now we don't apply this rule
         // for audio.
         if (!audio_configured_ && !last_packet_marker_bit_) {
           break;
         }
-        ssrc = ssrc_;
+        if (!ssrc_) {
+          LOG(LS_ERROR)  << "SSRC unset";
+          return 0;
+        }
+
+        ssrc = *ssrc_;
+
         sequence_number = sequence_number_;
         ++sequence_number_;
         payload_type = payload_type_;
         over_rtx = false;
       } else {
         // Without abs-send-time or transport sequence number a media packet
         // must be sent before padding so that the timestamps used for
         // estimation are correct.
         if (!media_has_been_sent_ &&
             !(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
               (rtp_header_extension_map_.IsRegistered(
                    TransportSequenceNumber::kId) &&
                transport_sequence_number_allocator_))) {
           break;
         }
         // Only change change the timestamp of padding packets sent over RTX.
         // Padding only packets over RTP has to be sent as part of a media
         // frame (and therefore the same timestamp).
         if (last_timestamp_time_ms_ > 0) {
           timestamp +=
               (now_ms - last_timestamp_time_ms_) * kTimestampTicksPerMs;
           capture_time_ms += (now_ms - last_timestamp_time_ms_);
         }
-        ssrc = ssrc_rtx_;
+        if (!ssrc_rtx_) {
+          LOG(LS_ERROR)  << "RTX SSRC unset";
+          return 0;
+        }
+        ssrc = *ssrc_rtx_;
         sequence_number = sequence_number_rtx_;
         ++sequence_number_rtx_;
         payload_type = rtx_payload_type_map_.begin()->second;
         over_rtx = true;
       }
     }
 
     RtpPacketToSend padding_packet(&rtp_header_extension_map_);
     padding_packet.SetPayloadType(payload_type);
     padding_packet.SetMarker(false);
@@ skipping 353 matching lines @@
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
   int avg_delay_ms = 0;
   int max_delay_ms = 0;
   {
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_)
+      return;
+    ssrc = *ssrc_;
   }
   {
     rtc::CritScope cs(&statistics_crit_);
     // TODO(holmer): Compute this iteratively instead.
     send_delays_[now_ms] = now_ms - capture_time_ms;
     send_delays_.erase(send_delays_.begin(),
                        send_delays_.lower_bound(now_ms -
                        kSendSideDelayWindowMs));
     int num_delays = 0;
     for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
@@ skipping 19 matching lines @@
   send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
 }
 
 void RTPSender::ProcessBitrate() {
   if (!bitrate_callback_)
     return;
   int64_t now_ms = clock_->TimeInMilliseconds();
   uint32_t ssrc;
   {
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_)
+      return;
+    ssrc = *ssrc_;
   }
 
   rtc::CritScope lock(&statistics_crit_);
   bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
                             nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
 }
 
 size_t RTPSender::RtpHeaderLength() const {
   rtc::CritScope lock(&send_critsect_);
   size_t rtp_header_length = kRtpHeaderLength;
@@ skipping 13 matching lines @@
                                 StreamDataCounters* rtx_stats) const {
   rtc::CritScope lock(&statistics_crit_);
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
 }
 
 std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
   rtc::CritScope lock(&send_critsect_);
   std::unique_ptr<RtpPacketToSend> packet(
       new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
-  packet->SetSsrc(ssrc_);
+  RTC_DCHECK(ssrc_);
+  packet->SetSsrc(*ssrc_);
   packet->SetCsrcs(csrcs_);
   // Reserve extensions, if registered, RtpSender set in SendToNetwork.
   packet->ReserveExtension<AbsoluteSendTime>();
   packet->ReserveExtension<TransmissionOffset>();
   packet->ReserveExtension<TransportSequenceNumber>();
   if (playout_delay_oracle_.send_playout_delay()) {
     packet->SetExtension<PlayoutDelayLimits>(
         playout_delay_oracle_.playout_delay());
   }
   return packet;
 }
 
 bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
   rtc::CritScope lock(&send_critsect_);
   if (!sending_media_)
     return false;
-  RTC_DCHECK_EQ(packet->Ssrc(), ssrc_);
+  RTC_DCHECK(packet->Ssrc() == ssrc_);

[stefan-webrtc, 2017/02/15 08:26:37]

DCHECK_EQ exists, right?

[nisse-webrtc, 2017/02/15 08:42:03]

Tried it earlier, but it doesn't support comparing an optional to a value.

   packet->SetSequenceNumber(sequence_number_++);
 
   // Remember marker bit to determine if padding can be inserted with
   // sequence number following |packet|.
   last_packet_marker_bit_ = packet->Marker();
   // Save timestamps to generate timestamp field and extensions for the padding.
   last_rtp_timestamp_ = packet->Timestamp();
   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
   capture_time_ms_ = packet->capture_time_ms();
   return true;
@@ skipping 11 matching lines @@
     return false;
 
   *packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
 
   if (!packet->SetExtension<TransportSequenceNumber>(*packet_id))
     return false;
 
   return true;
 }
 
-void RTPSender::SetSendingStatus(bool enabled) {
-  if (!enabled) {
-    rtc::CritScope lock(&send_critsect_);
-    if (!ssrc_forced_) {
-      // Generate a new SSRC.
-      ssrc_db_->ReturnSSRC(ssrc_);
-      ssrc_ = ssrc_db_->CreateSSRC();
-      RTC_DCHECK(ssrc_ != 0);
-    }
-    // Don't initialize seq number if SSRC passed externally.
-    if (!sequence_number_forced_ && !ssrc_forced_) {
-      // Generate a new sequence number.
-      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
-    }
-  }
-}
-
 void RTPSender::SetSendingMediaStatus(bool enabled) {
   rtc::CritScope lock(&send_critsect_);
   sending_media_ = enabled;
 }
 
 bool RTPSender::SendingMedia() const {
   rtc::CritScope lock(&send_critsect_);
   return sending_media_;
 }
 
 void RTPSender::SetTimestampOffset(uint32_t timestamp) {
   rtc::CritScope lock(&send_critsect_);
   timestamp_offset_ = timestamp;
 }
 
 uint32_t RTPSender::TimestampOffset() const {
   rtc::CritScope lock(&send_critsect_);
   return timestamp_offset_;
 }
 
-uint32_t RTPSender::GenerateNewSSRC() {
-  // If configured via API, return 0.
-  rtc::CritScope lock(&send_critsect_);
-
-  if (ssrc_forced_) {
-    return 0;
-  }
-  ssrc_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_ != 0);
-  return ssrc_;
-}
-
 void RTPSender::SetSSRC(uint32_t ssrc) {
   // This is configured via the API.
   rtc::CritScope lock(&send_critsect_);
 
-  if (ssrc_ == ssrc && ssrc_forced_) {
+  if (ssrc_ == ssrc) {
     return;  // Since it's same ssrc, don't reset anything.
   }
-  ssrc_forced_ = true;
-  ssrc_db_->ReturnSSRC(ssrc_);
-  ssrc_db_->RegisterSSRC(ssrc);
-  ssrc_ = ssrc;
+  ssrc_.emplace(ssrc);
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
 uint32_t RTPSender::SSRC() const {
   rtc::CritScope lock(&send_critsect_);
-  return ssrc_;
+  return *ssrc_;

[stefan-webrtc, 2017/02/15 08:26:37]

DCHECK that ssrc_ is set here first.

[nisse-webrtc, 2017/02/15 08:42:03]

Added here, and a couple of other places dereferencing ssrc_ or ssrc_rtx_.

 }
 
 rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
   if (video_) {
     return video_->FlexfecSsrc();
   }
   return rtc::Optional<uint32_t>();
 }
 
 void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
@@ skipping 59 matching lines @@
   // when transport interface would be updated to take buffer class.
   std::unique_ptr<RtpPacketToSend> rtx_packet(new RtpPacketToSend(
       &rtp_header_extension_map_, packet.size() + kRtxHeaderSize));
   // Add original RTP header.
   rtx_packet->CopyHeaderFrom(packet);
   {
     rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
       return nullptr;
 
+    RTC_DCHECK(ssrc_rtx_);
+
     // Replace payload type.
     auto kv = rtx_payload_type_map_.find(packet.PayloadType());
     if (kv == rtx_payload_type_map_.end())
       return nullptr;
     rtx_packet->SetPayloadType(kv->second);
 
     // Replace sequence number.
     rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
 
     // Replace SSRC.
-    rtx_packet->SetSsrc(ssrc_rtx_);
+    rtx_packet->SetSsrc(*ssrc_rtx_);
   }
 
   uint8_t* rtx_payload =
       rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
   RTC_DCHECK(rtx_payload);
   // Add OSN (original sequence number).
   ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
 
   // Add original payload data.
   auto payload = packet.payload();
@@ skipping 81 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc